IMPLEMENTATION with SIP Trunk (recommended)

The customer sends to LiveCall (help@livecall.io) external IP address of their PBX (call center software)
LiveCall then registers the IP address as “secure” (whitelisted)
LiveCall sends the customer its SIP Trunk data (IP, username, password, phone number).
The customer registers the SIP Trunk on their PBX.
Calls coming to the Trunk should be routed to a sales queue, preferably without an IVR and without long announcements.

SIP Specification:
|------------------|-----------------|
| Protocol | SIP |
|------------------|-----------------|
| IP for signaling | 31.186.82.127 |
| Port | 5060 |
|------------------|-----------------|
| IP for RTP | 31.186.82.127 |
| Port | 16384 - 32768 |
|------------------|-----------------|
| Codecs | G.711A |
|------------------|-----------------|
| DTMF | INBAND, RFC2833 |
|------------------|-----------------|
| CLIR SERVICE | SIP privacy |
| | RFC3325 |
|------------------|-----------------|

We use Freeswitch and RFC 3261 as a standard for SIP Communication.

If you can't register our SIP Trunk please contact us at help@livecall.io. We'll find a different solution.

IMPLEMENTATION without SIP Trunk

The customer defines the phone number which will be used to answer LiveCall callback requests (most cases there’s one number for customer support and one for sales).
The number should be routed to a sales queue and have no IVR.
Put this phone number to "Phone number" section when editing the call center User and tick "This number is a queue" checkbox.

If you have any questions or problems contact us at help@livecall.io or through chat.
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